I have a camera that takes videos at high FPS, and I can later use ffmpeg to slow them down for visual effect. For the video portion, it is possible to losslessly alter the frame rate - no transcoding necessary. Change one number from 120 to 24 and it plays exactly the same, but Slower.
Now, for audio. What I want is to slow / down-pitch the sound at the same rate to match. This is doable in .wav format (just tamper with the header). The camera shoots in AAC though. A transcode to WAV is technically "lossless", but it's also going to blow up the filesize. I can also re-encode back to AAC, but at a quality loss.
So, my question: Is it possible to "globally" modify some AAC header and alter its playback rate, without re-encoding? The plan would be to extract the AAC stream, twidle a couple bits to change the playback rate, and then re-embed it. Multimedia.cx seems to indicate this is possible (https://wiki.multimedia.cx/index.php/MPEG-4_Audio), but are there other considerations - e.g. tables that only work at some frequencies, or LZ compression that may refer back to this byte position, or whatever?
https://hydrogenaud.io/index.php/topic,33993.0.html
Playing at half sample rate might be possible, but I've never seen anyone actually do it.
Of the tests I've seen, transcoding with AAC isn't a perceptible problem; nothing at all like the wise internet parrots make it out to be.